Hearing aid apparatus

ABSTRACT

The acoustical-mechanical disturbance feedback between the electrical-acoustical converter and the acoustical-electrical converter of a hearing aid apparatus is compensated by means of an adaptive compensator filter which feeds back a signal derived from the output of an amplification filter to its input. At the input side thereby the signal from the acoustical-to-electrical converter and the output signal of the adaptive compensator filter are substracted at a difference forming unit, the output of which being led to the input of the amplification filter. The difference is thereby formed in time domain, and time domain to frequency domain transform is performed at the output side of the difference forming unit, accordingly inverse frequency domain to time domain transform at the electric input side of the electrical-to-acoustical converter.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention is generally directed to hearing aid technology,more specifically the present invention deals with problems which occurdue to acoustical-mechanical feedback from an electrical-to-acousticalconverter of hearing aid apparatus to its acoustical-to-electricalconverter.

2. Description of Prior Art

The problems which occur due to acoustical-mechanical feedback betweenthe electrical-to-acoustical converter--EAC--and theacoustical-to-electrical converter--AEC--of hearing aid apparatus areknown and are e.g. described in the EP-A-0 415 677 according to the U.S.Pat. No. 5,259,033 which documents shall form an integral part of thepresent description with respect to the mentioned problems.

An attempt to resolve these problems is schematically shown in FIG. 1which shows a prior art hearing aid apparatus.

Definition

Throughout the present description, two points of an electric circuitare considered to be "operationally connected" whenever an electricsignal at one of these two points is dependent from the electric signalat the second of these points, This irrespective of whether a directconnection of the two points is installed or whether the electric signalbetween the two points is led through signal treating units which changethe signal transmitted from the first to the second point. Such changesmay be amplification, filtering, superposition, time domain to frequencydomain transform, frequency domain to time domain inverse transform etc.

According to FIG. 1, a prior art hearing aid apparatus comprises an AEC1, the output of which being operationally connected to the input of ananalog-to-digital converter--ADC--3. A digital amplification filter unit5 is operationally connected with its output to a digital-to-analogconverter--DAC--7, which latter is operationally connected with itsoutput to the input of EAC 9.

With the block 11 in FIG. 1, the acoustical mechanical disturbancefeedback is shown with a transmission characteristic h, which isgenerally varying in time. The feedback signal y(t) is superimposed tothe acoustical signal v(t) to be amplified by the hearing aid apparatus.The superposition result acts on the input of the AEC 1, which, at itsoutput, generates the signal d(t) in time domain as a basis forgenerating time discrete sampling values d(nT) at the ADC 3 with timeintervals nT.

For suppression of the disturbing feedback signal y(t), e.g. in D. K.Bustamante et al., "Measurement and adaptive suppression of acousticfeedback in hearing aids", Proc. 1989, IEEE, ICASSP, 3:2017-2020, 1989,it has been proposed to provide a difference forming unit 13 and acompensator filter unit 15. The compensator filter unit 15 generatesfrom the output signal of the amplification filter unit 5, by means offiltering with an m-stage finite impulse response filter, an estimatesignal y(nT), which is fed to the difference forming unit 13. Thereby,making use of the well-known "least mean square" algorithm, thecoefficients of the filter of compensator filter unit 15 are iterativelyadjusted, so that the difference signal e(nT) at the output ofdifference forming unit 13 becomes not anymore correlated with theestimate signal y(nT). The compensator filter unit 15 thereby comprisesan adaption control input A to which the signal e(nT) is fed foradaption control of the filter coefficients.

Under presumption of uncorrelated signal v(t), thus of v(nT)(digitalized), and of the amplified signal u(t) and thus of u(nT) at theoutput of amplifier filter unit 5, which is reached by appropriateselection of the time-lag DT at the digital amplifier filter of the unit5, it becomes possible to rise the gain of the amplifier filter unit 5by 6 to 10 dB compared with such gain at a hearing aid apparatus withoutadaptive compensator filter unit 15.

Nevertheless, this approach has the drawback that, with an assuredlength of the filter of adaptive compensator filter unit 15 of m-stages,a number of 2 m multiplication operations per sample at the ADC 3 arenecessary. This leads to a very bulky system, especially considering theminiaturization which is necessary for hearing aid apparatusimplementation.

At the system shown in FIG. 1, it is further necessary that the stepwidth μ of the LMS algorithm is kept as small as possible to achievespeed signal transmission, so that adaption of the adaptive compensatorfilter unit 15 to the disturbance feedback 11 becomes accordingly slow.It follows therefrom that the possible increase of gain at the amplifierfilter unit 5 is restricted due to stability limits.

As an improvement of this known approach, according to FIG. 1, a furtherattempt was to couple into the system a stationar measuring signal as isknown e.g. from "Feedback cancellation in hearing aids: Results from acomputer simulation", J. M. Kates, IEEE, Trans.on Signal Processing,Vol. 39, No, 3, March 1991, or from the EP-A-0 415 677 (U.S. Pat. No.5,259,033). As a stationar measuring signal, a noise signal was coupledinto the system.

It is a drawback of this improved approach that a generator for themeasuring signal must be provided with an amplitude control to ensure asufficient signal-to-noise ratio.

By the last mentioned attempt and with a compensator filter of 32ndorder, an increase of gain at the amplifier filter unit 5 byapproximately 17 dB became possible.

Due to the drawbacks of the last mentioned attempt with measuring signalcoupling, a further approach as shown in FIG. 2 became known, accordingto "Integrated Frequency-Domain Digital Hearing Aid With the LappedTransform", S. M. Kuo and S. Voepel, Electronics Letters, vol. 28, no.23, November 1992.

According to this approach, the signal treatment is performed in thefrequency domain at the amplifier filter unit 5 and at the adaptivecompensator filter unit 15, according to FIG. 1. The output signal ofthe ADC 3 is transformed from time domain into frequency domain by meansof an overlapping orthogonal transform (LOT) at a transform unit 17. Anaccording inverse transform (ILOT) at an inverse transform unit 19generates for the input of the EAC 7 the time domain signal u(nT) asnecessary.

Because, when selecting a suitable time domain to frequency domaintransformation, especially the discrete Fourier Transform (DFT) or thediscrete Hartley Transform (DHT), the convolution at the adaptivecompensator filter unit 15_(f) and at the amplifier filter unit 5_(f),when transiting into the frequency domain becomes a multiplication, thisapproach results principally in a reduction of calculation effort, andthus of hardware installation. Nevertheless, structuring of the discretesignal d(nT) at the input of the transform unit 17 into blocks ofpredetermined length is necessary. Thereby the errors due to such blockseparation and compared with conventional convolution may not beeliminated with a lapped block separation at the apparatus as shown inFIG. 2. Such errors lead to a time varying system, even then, when thedisturbance feedback h and thus the adaptive compensation filter unit15_(f) would be considered to be time invariant. The system remains timevariant even if the disturbing feedback and its compensation are frozen.

Therefore, a compromise had to be made by selecting long block lengthsof e.g. 512 sampling values. This led to an inefficient compensation viathe adaptive compensation filter unit 15_(f). Accordingly, thepracticable gain increase at the amplifier filter unit 5_(f) reins below10 dB.

SUMMARY OF THE INVENTION

It is an object of the present invention to provide a hearing aidapparatus which

keeps the advantages of signal treatment in the frequency domain,

ensures time invariance of the system at a time varying disturbingfeedback,

allows to minimalize calculation and hardware installation to such anextent that signal treatment may be performed under the restrictedvolume conditions when realizing hearing aid apparatus.

This object is resolved by the hearing aid apparatus which comprises

an acoustical-to-electrical--AEC--converter with an output,

an electrical-to-acoustical--EAC--converter with an input.

an analog-to-digital--ADC--converter with an input operationallyconnected to the output of the AEC and with an output,

a digital-to-analog--DAC--converter with an output operationallyconnected to the input of the EAC,

a difference forming unit with a first and with a second input and withan output, the first input being operationally connected to the outputof the ADC,

an amplifier filter unit with an input and with an output, the inputbeing operationally connected to the output of the difference formingunit, the output being operationally connected to the input of the DAC,

an adaptive compensator filter unit with an input and with an output andwith an adaption control input, the input being operationally connectedto the output of the amplifier filter unit, the output beingoperationally connected to the second input of the difference formingunit, the adaption control input being operationally connected to theoutput of the difference forming unit,

a first transform unit with an input and with an output beingoperationally interconnected between the adaption control input and theoutput of the difference forming unit,

a second transform unit with an input and with an output beingoperationally interconnected between the input of the adaptivecompensator filter unit and the output of the difference forming unit,

an inverse transform unit with an input and with an output operationallyinterconnected between the output of the adaptive compensator filterunit and the second input of the difference forming unit,

the first and second transform units performing a fast orthogonaltransformation on time domain input signals to generate frequency domainoutput signals, the inverse transform unit performing a transforminverse to that of the transform units.

By the fact that the time domain to frequency domain transform is notanymore, as shown in FIG. 2, performed at the input side of thedifference forming unit 13_(f), but the difference at this unit isformed still in the time domain, the required time invariance of thesystem may astonishingly be established. Especially when selectingsuitably lapped block separation, it becomes possible to realize thetime domain to frequency domain transforms with significantly smallerblock lengths, which consequently improves efficiency of thecompensation filter action. This further allows to rise the gain at theamplifier filter unit 5_(f) drastically compared with the system of FIG.2.

Other objects, advantages and preferred features of the inventivehearing aid apparatus will become evident to the man skilled in this artwhen reading the description and the claims of the present application.

BRIEF DESCRIPTION OF THE DRAWINGS

The present invention, under all its aspects, will be better understoodand objects other than those set forth above will become apparent to theman Skilled in this art when consideration is given to the followingdetailed description thereof.

Such description makes reference to the annexed drawings, wherein:

FIG. 1 shows a simplified functional block diagram of a prior arthearing aid apparatus at which signal treatment occurs in the timedomain;

FIG. 2 shows in a representation in analogy to that of FIG. 1, a furtherprior art hearing aid apparatus at which signal treatment occurs in thefrequency domain at a feedback compensator and at an amplificationfilter according to FIG. 1;

FIG. 3 shows in analogy to FIGS. 1 and 2 a first embodiment of a hearingaid apparatus according to the present invention;

FIG. 4 shows a further preferred embodiment of the inventive hearing aidapparatus, based on that of FIG. 3, and shown in an analogrepresentation as FIGS. 1 to 3;

FIG. 5 shows a further preferred embodiment of the inventive hearing aidapparatus in a representation in analogy to that of the FIGS. 1 to 4which hearing aid apparatus is an improvement of that shown in FIG. 4;

FIG. 6 shows by means of a simplified signal flow/functional blockdiagram a preferred realization form of a transform unit which isprovided at the adaption control input and at the input of theamplification filter unit as realized at the embodiment of FIG. 5;

FIG. 7 shows by means of a simplified signal flow/functional blockdiagram a preferred embodiment of the amplification filter unit at aninventive hearing aid apparatus according to FIG. 5;

FIG. 8 shows a simplified signal flow/functional block diagram of apreferred realization of an adaptive compensation filter unit at theinventive hearing aid apparatus according to FIG. 5;

FIG. 9 shows by means of a simplified signal flow/functional blockdiagram the generation of a step width signal as a function of monitoredsignal power, whereby the step width signal, as formed preferably asshown in FIG. 9, is applied to the adaptive compensation filter unitaccording to FIG. 8;

FIG. 10 shows by means of a simplified signal flow/functional blockdiagram a unit which is preferably implemented when realizing theadaptive compensation filter unit as shown in FIG. 8;

FIG. 11 shows, departing from an inventive hearing aid apparatus asshown in FIG. 4, an embodiment as today preferred, shown in functionalblock diagram representation;

FIG. 12 shows a part of an improved embodiment of the inventive hearingaid apparatus according to FIG. 11 with modelling of the EAC in the timedomain and/or in the frequency domain;

FIG. 13 shows a functional block/signal flow diagram of an electricalmodelling unit, modelling the behaviour of a loudspeaker in time domainand as it is preferably implemented at the inventive hearing aidapparatus according to one of the FIGS. 3, 11 or 12 for modellingtransfer behaviour of the EAC of the hearing aid apparatus;

FIG. 14 shows, departing from the embodiment of FIG. 12, a furtherimprovement of a part of the inventive hearing aid apparatus at whichmodelling and/or amplitude limitation and/or the gain are controlled infunction of the instantaneous conditions of a battery feeding theinventive apparatus;

FIG. 15 shows, departing from the embodiment of FIG. 11, a furtherimprovement of the inventive hearing aid apparatus which resides in acontrolled appliance of a noise signal in frequency or in time domainand preferably selectively controlled;

FIG. 16 shows a preferred realization form of noise implementationaccording to FIG. 15 in the time domain;

FIG. 17 shows a preferred realization form of noise implementationaccording to FIG. 15 in the frequency domain.

DESCRIPTION OF PREFERRED EMBODIMENTS OF THE INVENTION

FIG. 3 shows by means of a signal flow/functional block diagram aprinciple of the present invention under a first aspect. The referencenumbers, which were already used in FIGS. 1 and 2 for functional blocksand signals, are also used in FIG. 3 to facilitate cross reference.

In the embodiments of the inventive apparatus according to both FIGS. 3and 4, the time discrete difference signal r(nT) is formed at thedifference forming unit 13 from the digitalized output signal d(t) ofthe AEC 1 and from the output signal of the adaptive compensation filterunit 15_(f). It is the time discrete difference signal r(nT) at theoutput of the difference forming unit 13 which is subjected to anoverlapping orthogonal transform LOT.

According to FIG. 3, the difference signal r(nT) is transformed by a LOTtransform unit 20 in the adaption control signal E[k] which is led tothe adaption control input A_(f) of the adaptive compensator filter unit15_(f). Because the time domain to frequency domain transform occurs atthe LOT-transform unit 20 with data blocks with a predetermined numberof samples from the difference signal r(nT), the feature [k] defines thenumber of a signal block at the output of the transform unit 20.

The difference signal r(nT) is fed according to FIG. 3 in time domain tothe amplification filter unit 5, the output thereof being operationallyconnected to the EAC 9 via the DAC 7. At the input, the DAC 7 receivesthe time discrete output signal u(nT) from the amplification filter unit5. This output signal u(nT) is subjected to a further orthogonaltransform at the transform unit 22, where it is transformed from timedomain into frequency domain. The output signal of the transform unit 22is fed to the signal input E_(f) of the adaptive compensator filter unit15_(f). The output signal Y[k+1] of the adaptive compensation filterunit 15_(f) is inverse transformed at an inverse transform unit ILOT 24from frequency domain back into time domain. The output signal y(nT) ofthe inverse transform unit 24 is led, as a time discrete signal, to thedifference forming unit 13.

Additionally to the embodiment according to FIG. 3 and now according toFIG. 4, not only signal treatment at the adaptive compensation filterunit 15_(f) is performed in the frequency domain, but also signaltreatment at the amplification filter unit 5_(f). Thereby a transformunit LOT 28 is provided, the frequency domain output thereof beingoperationally connected to the input of the amplification filter unit5_(f). An inverse transform unit ILOT 26 is operationally connected withits output to the input of the DAC 7. Compared with the embodiment ofFIG. 3, the embodiment of FIG. 4 has no transform unit 22.

Principally, and as was shown at the embodiments of the FIGS. 3 and 4,the main difference to prior art embodiments according to FIG. 2 is thatdifference formation at the difference forming unit 13 inventivelyoccurs in the time domain whereby the above mentioned drawbacks of priorart embodiments with respect to time variance become remedied.

Thereby, it becomes possible to deal with drastically reduced blocklengths at the LOT transform units 20, 22, 28 and accordingly at theinverse transform units 24 and 26 compared with the prior art approachaccording to FIG. 2. In a preferred embodiment of the present invention,the block length of the blocks numbered k is 128 samples.

FIG. 3 further shows an embodiment in which one transform unit LOT 20and one transform unit LOT 22 are respectively provided at the inputE_(f) of the adaptive compensation filter unit 15_(f) and at itsadaption control input A_(f).

A preferred embodiment is nevertheless that according to FIG. 4, inwhich a transform unit LOT 20 is provided for the adaption control inputA_(f) and a transform unit LOT 28 is provided with its outputoperationally connected to the input of the amplification filter unit5_(f). Thereby an inverse transform unit ILOT 26 is operationallyconnected with its output to the DAC 7.

It is known that for the formation and the treatment of data blocks inoverlapping orthogonal transforms principally two simple techniques areavailable, namely that of "overlap-save" and that of "overlap-add". Withrespect to these techniques, reference is made to the respectiveliterature as e.g. to "Signal processing with lapped transforms",Henrike S. Malvar, Artec House, Boston, 1992.

In a preferred embodiment of the present invention, and as shown in FIG.4, a LOT transform unit 28 is also provided at the input of theamplification filter unit 5_(f), an inverse transform unit 26 isprovided at the input of the DAC 7 and a further ILOT inverse transformunit 24 is provided at the output of the adaptive compensation filterunit 15_(f).

These transform and inverse transform units 38, 24, 26 operate in apreferred embodiment according to the "overlap-save" technique. Thereby,and in this preferred embodiment, the LOT transform unit 20 provided atthe adaption control input A_(f), according to FIG. 4, operatesaccording to the "overlap-add" technique.

This last mentioned preferred embodiment and block treatment lead to afurther preferred embodiment of the inventive hearing aid apparatus,which is shown in FIG. 5.

In opposition to the embodiment of FIG. 4, the time discrete differencesignal r(nT) is here operatively connected to a single LOT transformunit 30 from the output signal of which the adaption control signal E[k]fed to the adaption control input A_(f) as well as the input signal R[k]fed to the input of the amplification filter unit 5_(f) are derived.

As was mentioned, the overlapping orthogonal transform preferably baseson DFT.

FIG. 6 shows a realization form of a data transfer path of the timediscrete difference signal r(nT) at the output of the difference formingunit 13 to the adaption control input A_(f) as the adaption controlsignal E[k] and further to the input of the amplification control unit5_(f), as input signal R[k] according to FIG. 5.

According to FIG. 6, the output of the difference forming unit 13 withthe time discrete difference signal r(nT) is operationally connected tothe input of an overlap orthogonal transform unit 30a, which operates onthe basis of DFT. The transform unit 30a operates according to"overlap-add" technique as is marked in FIG. 6 by the "OA" index.Thereby at the input of the transform unit 30a, the error block e[k] isformed by dividing r(nT) into partial blocks with a length N. In apreferred embodiment the length is N=64. These blocks are lengthened toan overall length of 2N by hulling, thus, in the preferred embodiment,to the length of 2N=128. This means:

    e[k]=(0 . . . 0,r((k+1)NT), r((k+1)NT+T) . . . r((k+2)NT-T)).sup.T.

Its DFT, i.e. the signal E[k], is fed, according to a preferredembodiment according to FIG. 5, directly to the adaption control inputA_(f) of the adaptive compensation filter unit 15_(f). Via a time-lagunit 32, wherein a respective buffering occurs, subsequent data blocks,i.e. with the numbers k and k+1, are prepared. A superposition of theblocks, block partition by block partition, results directly in a blockR[k], now of the "overlap-save" type, which is directly led to the inputof the amplification filter unit 5_(f), as was previously mentioned aspreferred technique, according to FIG. 5. The superposition at the unit34 is thereby defined by

    R.sub.j [k]=E.sub.j [k]+(-1).sup.j E.sub.j [k-1],

wherein j (running from 0 to 2N-1) designates the number of therespective block partition.

By this realization a substantive reduction of hardware and calculationefforts are realized.

According to FIG. 7, the amplification filter unit 5_(f), which receivedthe data blocks R[k], comprises first an amplification filter 40, theoutput of which being operationally connected to the input of a time-lagunit 42 performing according buffering. Thereby, the parameter ddesignates the overall time-lag of the system considered from the outputof the ADC 3 to the input of the DAC 7 and normalized with the overlapparameter of the partial block length N. Due to this block treatment,there results a minimal time-lag of N samples according to a minimald-value of 1. In the preferred embodiment with a partial block length ofN=64 and with an overall block length of 2N=128, d was set on a value of2, thereby making use of a single partial compensator as will beexplained with reference to FIG. 8.

The block signal U[k+1] at the output of the time-lag unit 42 and of theamplification filter unit 5_(f) is operatively connected on one hand tothe input E_(f) of the adaptive compensator filter unit 15_(f) and onthe other hand to the input of the ILOT inverse transform unit 26, whereit is subjected to an inverse DFT transform in "overlap-save" technique.Because the resulting time signal u(nT) is generated with a time-lagaccording to a partial block length N, the block numbering k+1 of thesignal U[k+1] is justified.

In FIG. 8 a preferred embodiment of the adaptive compensation filterunit 15_(f) at the inventive hearing aid apparatus according to FIG. 5is shown. Thereby, block signals U[k+1] to U[k+1-L] are prepared bybuffering with time-lag units of the type as shown at 56. Therefrom, andwith the help of partial compensators, the first of which being definedby the reference number 50, partial estimate signals Y₁ [k+1] to Y_(L)[k+1] are generated, which partial estimate signals are added at anaddition unit 52 to result in the overall estimate signal Y[k+1]. Asshown in FIG. 5, there occurs subsequently in the ILOT inverse transformunit 24 the inverse transform back into time domain, in the preferredembodiment by means of an inverse DFT transform of "overlap-save" type.

With reference to the first partial compensator, the partial estimatesignal Y₁ [k+1] appears at the output of the multiplication unit 64,whereby the block signals U[k+1] and the block weighing signal H₁ [k+1]are applied to the inputs of the multiplication unit 64. Themultiplication is thereby performed for each block partition accordingto the formula

    Y.sub.i,j [k+1]=U.sub.j [k+2-i]H.sub.i,j [k+1],

wherein j designates the block partition from 0 to 2N-1 and i designatesthe number of the partial compensator considered, from 1 to L.

The block weighing H_(i) [k+1] represents thereby the actual estimate inthe frequency domain for the partition i of the length N of the timediscrete pulse response h of the acoustical-mechanical disturbancefeedback 11. The estimate H_(i) [k+1] is actualized on the basis of theformer estimate H_(i) [k] previous to the formation of Y_(i),j [k+1]. Todo so, and again with reference to the first partial compensator, theblock signal U[k+1-1] and the step width μ[k+1-1] are fed to themultiplication unit 54, the output signal of which being fed to themultiplication unit 58 together with the block signal E[k]. The outputof multiplication unit 58 is then used for actualizing H₁ [k+1] in thesummation unit 60, according to formula

    H.sub.i,j [k+1]=H.sub.i,j [k]+μ.sub.j [k+1-i]U*j[k+1-i]E.sub.j [k].

The index (*) stands for "conjugate complex number", j designates againthe block partition and i the partial compensator.

A realization by means of partial compensators has the advantage thatthe minimal time-lag D=N may be adjusted independently from the lengthof the pulse response of the disturbance feedback 11 by appropriateselection of the partial block length N. Thereby, a "trade-off" betweentime-lag D and the partial block length N, which determines theefficiency of operation, becomes possible. Further, specific parts ofthe pulse response h may be specifically influenced by according blockweighing in the frequency domain, e.g. according to the acoustical low-and long-distance parts.

Principally, each known method may be used for governing the step widthμ[k].

In FIG. 9 an embodiment preferred today is shown for generating thenormalized step width μ[k] according to FIG. 8, which may additionallybe used for disabling the adaption procedure. Thereby, and e.g.departing from the block signal U[k] according to FIG. 8, this blocksignal is used to calculate the actual block signal μ[k] before it isapplied to the multiplication unit 54. This is done in that the blocksignal U[k] is led to a signal-power determining unit 70 which acts withits output onto two interpolation filters 72 and 74. The interpolationfilters 72 and 74 control with their outputs the scaling unit 78, whichgenerates the scaling value S[k] led to the input of the multiplicationunit 80. The scaling value S[k] is used for normalizing the referencestep width μ₀.

The interpolation filters operate according to the formula

    P.sub.U,j [k]=c(1-γ)U*j[k]U.sub.j [k]+γP.sub.U,j [k-1]

and are parametrisized with γ and c. The index j stands for the blockpartition. In the preferred realization form γ was selected to be 0.8and c=1 for filter 72 and γ=0.995 and c=0.2 for the filter 74.

If for the interpolator filter 74 γ is selected to be 1, then thisinterpolator filter is void and it remains a block signal P_(U) ^(min)which is constant in time and which may suffice in some cases, furtherreducing hardware and calculation efforts.

The scaling value S[k] is on one hand used for normalizing the referencestep width μ₀ via the output of the filter 72 which is referred to inFIG. 9 by the block signal P_(U) [k]. On the other hand, the scalingvalue S[k] is used to freeze or disable the adaption procedure ofspecific frequency components via the output of the filter 74 which isdesignated in FIG. 9 as block signal P_(U) ^(min) [k], if efficiency isnot satisfying. Thereby the scaling value S[k] is formed according toformula ##EQU1## whereby the j again designate the block partition.

In FIG. 10 there is shown a further preferred embodiment whichsignificantly improves the speech quality when partial compensatorsaccording to FIG. 8 are used and at unchanged further parameters.Thereby the estimate H_(i) [k+1] of the partial compensator i is ledpreviously to multiplication with U[k+2-i] at the multiplication unit 64of FIG. 8 to a projection unit 62. E.g. the block weight H_(i) [k+1] isthereby subjected to an inverse DFT transform at unit 82 and is thencleaned, by nulling all block partitions with the indexes N to 2N-1 atthe unit 84. Finally, the output signal of unit 84 is back-transformedinto the frequency domain by the DFT unit 86.

As is known, the EAC 9 is not linear in the sense that it does notanymore linearly transform the input signal into an output signal if theinput signal is larger than a predetermined input signal level. Besidesthe acoustical distortions which are caused by such behaviour, it mustbe considered that the signal transmission path via the adaptivecompensation filter unit 15_(f) should be adapted as exactly as possibleto the signal path via the functional blocks 7, 9, 11, 1 and 3. Theadaptive compensation filter unit as described up to now may not takeinto account such non-linearities. Additionally, the maximum acousticaloutput level of the hearing aid apparatus should be adjustable accordingto individual needs of the users, Thereby the problem that the converter9 could be driven in its non-linear operating range does obviously onlyoccur if the individually adjusted maximum output level my still drivethe converter 9 in the said non-linear operating range.

Based on these considerations and in a further preferred embodiment,also under a more general separate aspect of the present invention andas shown in dashed lines in FIG. 3, a limiter unit 90 operating in thetime domain in the specific embodiment according to FIG. 3 is providedat the output of the amplification filter unit 5. This limiter unit 90limits the output signal amplitude from the amplification filter unit 5,so that the EAC converter 9 is never driven in its non-linear operatingrange. Additionally, the limiting unit 90 enables to individually setthe maximum output level of the acoustic signal at EAC 9 as isschematically shown with the double arrows in block 90.

At the embodiment according to FIG. 4, the aspect of individual maximumpower setting and of not linear operation of EAC 9 are considered byproviding at the output of the amplifier filter unit 5_(f), whichoperates in the frequency domain, a unit 90_(f), which, in the frequencydomain, limits the frequency components of the signal spectrumconsidering their respective phasing, so that at the output of theinverse transform unit 26 and of the DAC 7 a time varying signal u(t) isformed which never drives the EAC 9 into its not linear operating range.Unit 90_(f) additionally allows to set or adjust individually a maximumoutput level for the EAC 9.

The same technique is realized with the unit 90_(f) at the embodiment ofthe invention according to FIG. 5.

FIG. 11 shows a further preferred embodiment of the inventive hearingaid apparatus which generally accords with the embodiment according toFIG. 4 with the difference that the inverse transform unit 26 accordingto FIG. 4 appears, according to FIG. 11, as unit 26a directly at theoutput of the amplification filter unit 5_(f). At the input of theadaptive compensation filter unit 15_(f), there is provided a LOTtransform unit 22a as was discussed above. In spite of the fact that theembodiment of FIG. 11 does not seem to be of any advantage compared withthe embodiment of FIG. 4, the embodiment of FIG. 11 allows to realizeoptions which are discussed below.

As may be seen from FIG. 4, which shows, as does FIG. 5, a preferredembodiment of the inventive hearing aid apparatus, provision of alimiter unit is only possible in the frequency domain because such alimiter unit must be effective in the feedback compensation signal pathwith the adaptive compensation filter unit 15_(f) too.

As now may be seen from FIG. 11, the functional block structure hereallows to provide the limiter unit 90 operating in the time domain whichleads to a limiter unit 90 which is significantly simpler to realizecompared with a limiter unit operating in the frequency domain.

This also allows introduction of further improvements by units forcompensating not linear effects as described in the following.

For ensuring an accurate identification of the EAC 9 by the adaptivecompensation filter unit 15_(f), first the input level to the EAC 9 islimited to prevent that this converter 9 is operated in its not linearsaturation range. This leads to a reduction of the maximum possible gainof the inventive hearing aid apparatus between AEC 1 and EAC 9.

In FIG. 12 a preferred embodiment of signal treatment at the input sideof the adaptive compensation filter 15_(f) and at the output side of theamplification filter 5_(f) is shown for an improved embodimentprincipally according to the apparatus according to FIG. 11.

According to FIG. 12 the EAC 9 with its non-linearities is modelledprincipally in the signal path with the adaptive compensation filterunit 15_(f). This is realized by a modelling unit 92 at the input of thetransform unit 22a according to FIG. 11, which modelling unit 92 thusoperates in the time domain. Additionally or alternatively, a modellingunit 92_(f) may be provided at the output side of the transform unit22a, which thus operates in the frequency domain.

By this realization it is reached that, depending on the accuracy of themodelling unit 92, the limit set at the unit 90, and thus the limit forits output signal, may be risen by approximately 6 dB compared with theembodiment according to FIG. 11. Thereby, it is also possible to omitunit 90.

The modelling unit 92 may be e.g. realized as described in R. Isermann,"Identifikation dynamischer Systeme" (Identification of dynamicsystems), Springer Verlag, 2:238, 1988, as a simplified Wiener-Model.

The transform into time domain between amplification filter unit 5_(f)and adaptive compensation filter unit 15_(f) allows, additionally and aswas described before, the addition of a not linear correction filterinto the signal path with the amplification filter unit 5_(f).

This may be realized, as shown in FIG. 12, by means of a modelling unit94 at the output of the inverse transform unit 26a and thus operating inthe time domain and/or by a modelling unit 94_(f) at the input of theinverse transform unit 26a and thus operating in the frequency domain.

It is clearly possible to replace the LOT transform units 20 and 28 ofthe embodiments of FIGS. 11 and 12 by a single LOT transform unit 30 asshown in the FIGS. 5 and 6.

In FIG. 13 the realization of a modelling unit modelling the behaviourof a loudspeaker and thus of EAC 9 is shown, operating in the timedomain. Such modelling unit is considered per se as inventive.Especially with the hearing aid apparatus according to the presentinvention and according to FIGS. 3 and 11 such a modelling unit is usedas block 90 and, according to FIG. 12, instead of the blocks 92, 90, 94respectively.

The modelling unit comprises a prefilter 100 with a transfercharacteristic F₁ (ω) being substantially a low path characteristic. Thecorner frequency ω₁ of the Bode diagram schematically shown in prefilterblock 100 is approximately 0.8 kHz in a preferred embodiment, the gain|F₁ | at this corner frequency ω₁ approximately 0 dB. The slope S₁ isapproximately 0 dB/DK.

The identification entities, namely corner frequency ω₁ and the slopesS₁ and S₂ as well as the gain, e.g. at the corner frequency ω₁, arefound by identification of the loudspeaker or EAC 9 to be modelled.

Following the prefilter 100, there is provided a linear amplificationunit 102 at which the amplification factor K is set. Following thelinear amplification unit 102, there is provided a not-linearamplification unit 104. The transfer characteristic of the not-linearamplification function Y=Q(x) is e.g.:

    y=x+ax.sup.2 +bx.sup.3 +cx.sup.4 +dx.sup.5.

For small input signals, the amplification of the not-linearamplification unit 104 is unity, so that the amplificationcharacteristic adjacent to the origin has the slope 1. For larger inputsignals x the not linear amplification characteristic has, as is knownfrom loudspeakers or from EAC 9, saturation characteristic.

The coefficients a, b, c, d of the not-linear amplificationcharacteristic and the amplification factor K are determined byidentifying the converter to be modelled.

Following the not-linear amplification unit 104, there is provided alinear amplification unit 106, whereat the amplification K of the linearamplification unit 102 is compensated, K⁻¹. Following the unit 106,there is provided a filter unit 108 substantially with high passcharacteristic, which, as is shown in FIG. 13, substantially compensatesthe frequency characteristic of the prefilter 100.

Thus, the converter modelling unit, i.e. the loud speaker or EAC 9modelling unit as shown in FIG. 13, comprises substantially a linearamplification part formed by the units 100, 102, 106, 108 and anot-linear amplifier unit 104.

Saturation and thus limiting phenomena may have, besides the two originsmentioned--namely wanted limitation of the maximum output level of EAC9, according to individual need, or driving EAC 9 into its converterspecific, not-linear saturation area--a third reason: It may be causedby a drop of battery voltage which supplies the inventive apparatus.Ageing of the battery which supplies the hearing aid apparatus leadsespecially at the DAC 7 to a decrease of signal gain and thus to adecrease of full-scale analog output signal.

Additionally, the output impedance of the battery appears normally inseries to the impedance of the EAC 9. Thus, with increased ageing of thebattery, the increasing battery output impedance, which appears inseries to the EAC 9, leads to an impedance at the output side of DAC 7which varies in time. This affects the non-linearities at the outputside of DAC 7 to be modelled as discussed above.

With the object of maintaining a high accuracy of the compensation ofthe disturbance feedback, as is the main object of the presentinvention, and to maintain stability of that compensation, the limitingunit 90, according to the FIGS. 3 or 4, operating in time domain, or90_(f), operating in the frequency domain, according to FIG. B, iscontrolled by the instantaneous battery output voltage and/or theintantaneous battery output impedance.

Departing from the embodiment of FIGS. 11 and 12, such battery statecontrol is schematically shown in FIG. 14. At the output of the batteryunit 120 which, as schematically indicated by "block powering",electrically supplies the electronic active components of the functionalblocks as described of the inventive hearing aid apparatus, there isprovided a monitoring unit 122 which monitors the instantaneous batteryoutput voltage U_(B) and/or the instantaneous battery output impedanceZ_(B). There result accordingly measuring signals e(U_(B)) and/ore(Z_(B)). These measuring signals control the limiting unit 90 and,analogically in the frequency domain, the limiting unit 90_(f) accordingto FIGS. 3, 4, 5, 11, 12 and 14, and/or the modelling units 92, 92_(f)or, respectively, 94, 94_(f) of FIGS. 12, 13, 14.

Preferably the measuring signals e are digitalized in that themonitoring unit 122 is operationally connected with an ADC (not shown inFIG. 14).

By the instantaneous battery output voltage and/or output impedance,especially the limits of the limiting units and/or the parameters of themodelling units are adjusted, thereby taking into account theinstantaneous battery state.

The parameters of modelling at the modelling units 92, 92_(f) or 94,94_(f) are adjusted in that they are changed by calculation as afunction of the said battery state or in that different sets of suchparameters are stored and are enabled by and according to theinstantaneous battery state.

As further shown in FIG. 14, a decrease of gain at the DAC 7 due to adrop of the battery output voltage may be compensated as a function ofthe measuring signal e: If the battery voltage drops and thereby thegain at the DAC 7, the measuring signal e controls the gain at block 7to be compensatorily increased. The battery voltage drop additionallyacts like a signal limitation by a limiter and is preferably consideredby means of a limiter unit 90_(b) at the input side of the modellingblock 92 or 92_(f) according to FIG. 14, which limiter unit 90_(b) iscontrolled by the instantaneous battery output voltage.

If a limiter unit 90_(b) according to FIG. 14 is provided, the units 90may be omitted. If modelling unit 92 or 92_(f) are provided, the units94 or 94_(f) may be omitted so that a relatively simple feedbackcompensation is reached, which is independent of the instantaneousbattery voltage.

On the ocher hand, the function of the unit 90_(b) may completely bereplaced by units 90 or 90_(f) according to FIGS. 4 or 5, which arecontrolled as a function of battery output voltage.

Taking the battery voltage drop into account with respect to signallimitation by means of limiter units as 90, 90_(f) or 90_(b) is of highimportance for ensuring stability of the hearing aid apparatus as thebattery voltage varies significantly.

For maintaining stability of feedback compensation, even in very loudsurroundings, where, e.g. according to FIG. 11, the AEC 1 could besaturated and thus would become not linear, there is provided, ifnecessary, a not linear model of the AEC 1, which, if necessary, alsomodels the behaviour of the ADC 3. Such modelling unit is providedbetween the output of the adaptive compensator filter unit 15 of FIG. 1or 15_(f), e.g. according to FIG. 11, and the substraction input of thedifference unit 13 e.g. according to FIGS. 1 or 11.

According to where such modelling unit is arranged, it operates in thefrequency domain or in the time domain, as is shown at 91 or 91_(f) inFIG. 11. For the modelling unit 91 or 91_(f) respectively, modelling thebehaviour of the AEC 1, the same considerations are valid which weredescribed with respect to modelling the EAC 9 by means of modellingunits 92, 92_(f). Provision of an AEC modelling unit, in fact of amicrophone model at a disturbance feedback compensated hearing aid,generally is considered one separate aspect of the present invention.The same is valid for a loudspeaker model as e.g. shown in FIG. 13.

A further improvement of effect of the adaptive compensation filter unit15_(f) may be reached in that a noise signal in time domain is infed asshown in FIG. 15 at the output side of the amplification filter unit5_(f).

This is realized, as shown in FIG. 15, in that a spectrum detector 125monitors the instantaneous signal spectrum at the output side of theamplification filter unit 5_(f) and e.g. monitors the significance ofpower peaks at specific frequency components, i.e. generally powerdensity distribution of the spectrum. If characteristic of the frequencyspectrum which is monitored at the unit 125 does not anymore fulfilpredetermined conditions, e.g. in that it leaves a predetermined powerdensity distribution, the unit 125 enables the output signal of a noisegenerator 127 to be superimposed at a superposition unit 129 to thesignal at the output side of unit 5_(f) in the form of digital noise r.To thereby reduce audibility of such noise, a filter unit 133 may beprovided at the output of the noise generator 127 as shown in FIG. 16,which filter unit forms the noise so that audibility of the superimposednoise is low enough compared with the audio signal at the output of EAC9, is e.g. lower by a level of 40 dB.

As is further shown at 131 in dashed lines in FIG. 15, the noise mayalso be fed to the inventive system in the frequency domain. If noise intime domain is introduced, then the noise generator 127 may e.g.comprise a BPRN. If noise is introduced in the frequency domainaccording to noise generator 127a of FIG. 17, then the noise generatorcomprises e.g. tables with noise spectra or a noise generatingalgorithm.

FIG. 16 shows, departing from the embodiment of FIG. 15, a preferredrealization of noise appliance in the time domain. The output signal ofthe amplification filter unit 5_(f) is monitored at a spectrum shapedetector unit 125_(a), If the spectrum shape leaves a predeterminedlimit characteristic, the output signal of the noise generator 127 issuperimposed via a linear filter 133 to the signal u(nT) according toFIG. 15 and, as is schematically shown, with the switching unit 135. Thenoise is preferably introduced at the input side of the limiter unit 90.As is further shown with a control line sc, the transfer characteristicof the filter 133 is preferably controlled in function of theinstantaneous spectrum at the input of the inverse transform unit 26a.

FIG. 17 shows a preferred realization form of noise appliance in thefrequency domain according to the dashed line representation at block131 of FIG. 15. The spectrum at the output of the amplification filterunit 5_(f) is again monitored at a spectrum shape detector unit 125_(b)in analogy to the unit 125_(a) of FIG. 16. The output signal of a noisegenerator 127a, wherein noise spectra are e.g. stored in tables and areselectively enabled, is superimposed to the spectrum at the output ofthe amplification filter unit 5_(f) via a spectrum shaping filter 137 asschematically shown by the switching unit 135_(a). This occurs wheneverthe spectrum shape detector unit 125_(b) detects a spectrum shape whichnecessitates superposition of noise.

The superposition of the noise signal in the frequency domain occurs atan addition unit 129_(a). The shaping filter 137 is again controlled bythe instantaneous spectrum, e.g. at the output of the amplificationfilter unit 5_(f), so as to ensure minimal audibility of the noisecoupled into the system.

Principally, introduction of noise controlled by the instantaneousspectrum of the signal transmitted from an AEC to an EAC of a hearingaid apparatus, as concerns its amplitude and/or spectral distribution,is considered a Separate object of the present invention.

We claim:
 1. Hearing aid apparatus, comprising:anacoustical-to-electrical--AEC--converter with an output, anelectrical-to-acoustical--EAC--converter with an input, ananalog-to-digital--ADC--converter with an input operationally connectedto the output of the AEC and with an output, adigital-to-analog--DAC--converter with an output operationally connectedto the input of the EAC, a difference forming unit with a first and witha second input and with an output, the first input being operationallyconnected to the output of the ADC, an amplifier filter unit with aninput and with an output, the input being operationally connected to theoutput of the difference forming unit, the output being operationallyconnected to the input of the DAC, an adaptive compensator filter unitwith an input, an output and an adaption control input, the input beingoperationally connected to the output of the amplifier filter unit, theoutput being operationally connected to the second input of thedifference forming unit, the adaption control input being operationallyconnected to the output of the difference forming unit, a firsttransform unit with an input and with an output being operationallyinterconnected between the adaption control input and the output of thedifference forming unit, a second transform unit with an input and withan output being operationally interconnected between the input of theadaptive compensator filter unit and the output of the differenceforming unit, an inverse transform unit with an input and with an outputbeing operationally interconnected between the output of the adaptivecompensator filter unit and the second input of the difference formingunit, said first and second transform units performing a fast orthogonaltransformation on input signals in time domain into output signals infrequency domain, said inverse transform unit performing a transformbeing inverse to that of the transform units.
 2. The apparatus of claim1, wherein the second transform unit is interconnected between theoutput of the amplifier filter unit and the input of the adaptivecompensation filter unit.
 3. The apparatus of claim 1, wherein thesecond transform unit is interconnected between the output of thedifference forming unit and the input of the amplifier filter unit and afurther inverse transform unit is operationally interconnected betweenthe input of the DAC and the output of the amplifier filter unit.
 4. Theapparatus of claim 3, wherein the first and the second transform unitsare formed by a single combined transform unit.
 5. The apparatus ofclaim 3, wherein at least the second transform unit, the one and thefurther inverse transform units operate in the overlap-save technique.6. The apparatus of claim 1, wherein the first transform unit operatesin the overlap-add technique.
 7. The apparatus of claim 4, wherein thecombined transform unit operates in the overlap-add technique and itsinput is operationally connected to the output of the difference formingunit, its output is operationally connected to the adaption controlinput and to a block storage unit, wherein, successively, successivedata blocks having been formed in the combined transform unit arestored, and further comprising an addition unit, wherein storagepartitions of the store, which accord to respective data blockpartitions, are added under consideration of the signal, the output ofthe addition unit providing data blocks of overlap-save type and beingoperationally connected to the input of the amplifier filter unit. 8.The apparatus of claim 1, wherein the amplifier filter unit comprises anamplifier filter and a time-lag unit, the output of the amplifier filterbeing operationally connected to the input of the time-lag unit.
 9. Theapparatus of claim 7, wherein the adaptive compensation filter unitcomprisesan input and a series of time-lag stages, the input of thefirst time-lag stage of the series being operationally connected to theinput of the adaptive compensation filter unit, 1≦i≦L partialcompensator units, wherein partial estimation signals

    Y.sub.i [k+1] for 1≦i≦L

are generated, wherein k stands for the number of data blocks counted atthe output of the combined transform unit, an addition unit, wherein thepartial estimation signals Y_(i) [k+1] generated by the L partialcompensators are added, the output of the addition unit being the outputof the adaptive compensator filter unit.
 10. The apparatus of claim 9,comprising a series of partial compensators, the input of the first ofthe series of partial compensators being operationally connected to theinput of the adaptive compensation filter unit and the input of eachpartial compensator of the series of partial compensators beingconnected to its output via a time-lag stage of the series of time-lagstages.
 11. The apparatus of claim 10, wherein each partial compensatorcomprises:a first multiplication unit with a first and a second inputand with an output, the first input being operationally connected withthe output of the partial compensator, a second multiplication unit witha first and with a second input and with an output, the first inputbeing operationally connected with the output of the firstmultiplication unit, the second input being operationally connected withthe adaption control input, whereby the output of the secondmultiplication unit is operationally connected via an accumulation unitto a first input of a third multiplication unit, the second inputthereof being operationally connected with the input of the partialcompensator, the output thereof being operationally connected to aninput of the addition unit of the adaptive compensation filter.
 12. Theapparatus of claim 3, wherein the output of the second transform unit isfurther operationally connected to the input of a signal powermonitoring unit, the output of which controlling the effect of a signalapplied to the adaption control input in dependency of whether thesignal power measured reaches or does not reach a predeterminedthreshold value.
 13. The apparatus of claim 11, wherein the second inputof the first multiplication unit is operationally connected to theoutput of a fourth multiplication unit with a first and a second input,to the first input of which a signal according to a reference step widthis fed, the second input thereof being operationally connected to theoutput of a scaling unit, which scaling unit being operationallyconnected at its inputs with the outputs of two interpolation filters,to which interpolation filters the output signal of the amplificationfilter unit is fed via a signal power measuring unit.
 14. The apparatusof claim 13, wherein, instead of the output signal of one of theinterpolation filters, a signal which is constant in time is fed to thescaling unit.
 15. The apparatus of claim 11, wherein an inversetransform unit, a hulling unit and a transform unit are interconnectedbetween the output of the accumulation unit and the first input of thethird multiplication unit.
 16. The apparatus of claim 1, wherein theoutput of an amplitude limiting unit is operationally connected to theinput of the EAC.
 17. The apparatus of claim 3, wherein an amplitudelimiting unit is operationally connected between the output of theamplifier filter unit and the input of the DAC.
 18. The apparatus ofclaim 1, wherein a modelling unit, modelling at least one of EAC and AECand operating in at least one of frequency and of time domain, isprovided at least one of operationally connected to the input and ofoperationally connected to the output of the adaptive compensationfilter unit, the modelling unit modelling the behaviour of the EACand/or AEC.
 19. The apparatus of claim 2, the output of an EAC- and/orAEC-modelling unit, modelling the EAC and/or AEC in the time domain, isoperationally connected to the input of the second transform unit. 20.The apparatus of claim 2, wherein an input of an EAC- and/orAEC-modelling unit, modelling the EAC and/or AEC in the frequencydomain, is operationally connected to the output of the second transformunit.
 21. The apparatus of claim 1, wherein one modelling unit,modelling the EAC and/or AEC, is provided with its output operationallyconnected to the input of the adaptive compensation filter unit andanother modelling unit, modelling the EAC and/or AEC, is provided withits input operationally connected to the output of the amplificationfilter unit.
 22. The apparatus of claim 21, wherein at least one of themodelling units operate in the time domain.
 23. The apparatus of claim1, further comprising at least one modelling unit, modelling thebehaviour of the EAC and/or of the AEC, the modelling unit comprising alinear transfer unit and a non-linear transfer unit.
 24. The apparatusof claim 23, wherein the linear transfer unit comprises at least oneamplifier and at least one filter.
 25. The apparatus of claim 24,wherein the linear transfer unit comprises a prefilter substantiallywith low pass characteristic, the output of which being operationallyconnected to the non-linear transfer unit, the output of the non-lineartransfer unit being operationally connected with a compensating filterunit with a frequency characteristic substantially inverse to thefrequency characteristic of the prefilter.
 26. The apparatus of claim25, wherein the output of a linear amplification unit is operationallyconnected to the input of the non-linear transfer unit and a linearamplification compensating unit is operationally connected with itsinput to the output of the non-linear transfer unit, which linearamplification compensating unit compensating amplification of the linearamplification unit.
 27. The apparatus of claim 1, comprising at leastone limiter unit operating in at least one of time domain and offrequency domain and an energy supply battery arrangement, furthercomprising a determining unit for determining the momentarily batterystate, the output of the determining unit controlling the at least onelimiter unit at a control input thereof.
 28. The apparatus of claim 1,wherein said DAC comprises a gain control input and comprising an energysupply battery, further comprising a determining unit for themomentarily state of the battery, the output of the determining unitbeing operationally connected to the gain control input of said DAC. 29.The apparatus of claim 1, comprising a modelling unit with at least oneparameter control input, modelling the behaviour of the EAC and/or theAEC, further an energy supply battery and a determining unit for themomentarily state of the battery, the output of the determining unitbeing operationally connected to the at least one parameter controlinput of the modelling unit.
 30. The apparatus of claim 29, wherein themodelling unit operates in time domain.
 31. The apparatus of claim 1,wherein the output of a noise generator is operationally connected tothe input of the adaptive compensation filter unit via a superpositionunit.
 32. The apparatus of claim 31, wherein the superposition iscontrolled.
 33. The apparatus of claim 31, wherein time-spans, duringwhich superposition occurs, are controlled.
 34. The apparatus of claim31, wherein the output signal of the noise generator is in the timedomain or in the frequency domain.
 35. The apparatus of claim 34,wherein the output of the amplification filter unit is operationallyconnected to the input of a shape detection unit, wherein theinstantaneous shape of input signal frequency spectrum is monitored andwherein a check is performed whether the instantaneous shape accordswith at least one predetermined condition or not, whereby the outputsignal of the shape detection unit controls the superposition.
 36. Theapparatus of claim 34, wherein the output of the noise generator isoperationally connected with the superposition unit via a shapingfilter, shaping amplitude and/or frequency distribution of the noise,shaping of the shaping filter being controlled by the instantaneousspectrum of the output signal of the difference forming unit.
 37. Ahearing aid apparatus, comprising:an acoustical-to-electricalconverter--AEC--, an electrical-to-acoustical--EAC--converter, anelectrical transmission circuit operationally connecting the output ofthe AEC and the input of the EAC, the circuit comprising a noisegenerator and a superposition unit at which a signal dependent from theoutput signal of the noise generator is superimposed to a signaldepending from a signal generated at the output of the AEC, the outputof the noise generator being operationally connected to thesuperposition unit via a filter unit with a control input for itstransmission characteristic, the control input being fed by a signaldependent on a signal generated at the output of said AEC, via afrequency spectrum monitoring unit.
 38. The apparatus of claim 37,wherein the noise generator generates a noise signal in time domain andthe filter unit is a linear filter unit.
 39. The apparatus of claim 37,wherein the noise generator generates a noise in frequency domain andthe filter is a spectrum shaping unit.
 40. A hearing aid apparatus,comprising:an acoustical/electrical converter--AEC--, anelectrical/acoustical converter--EAC--, an electrical transmissioncircuit operationally connecting the output of the AEC to the input ofthe EAC and comprising at least one transform unit performing fastorthogonal transform from time domain into frequency domain on anelectric signal dependent from a signal at the output of the AEC, anoise generator with an output, the output of the noise generator beingoperationally connected to a superposition unit at the frequency domainoutput side of the transform unit.